The Asterisk Development Team would like to announce the first release candidate of Asterisk 13.18.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.18.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Improvements made in this release:
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chan_sip: Provide access to read the full SIP Request-URI from INVITE (Reported by David J. Pryke) |
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alembic: Add support for Microsoft SQL server (Reported by Florian Floimair) |
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libsrtp-2.1.x support (Reported by Alexander Traud) |
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Enable CHANNEL function to get from and to tag from SIP Headers (Reported by Andre Nazario) |
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Google OAuth 2.0 support for XMPP / Motif (Reported by Andrey) |
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Support for GMIME 3.0 (Reported by Tzafrir Cohen) |
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app_queue: Add Priority to AMI QueueStatus (Reported by Niklas Larsson) |
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chan_pjsip: Port SIPDtmfMode to chan_pjsip (Reported by Torrey Searle) |
Bugs fixed in this release:
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res_pjsip: user=phone added to Anonymous caller-id when it shouldn’t be. (Reported by dtryba) |
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res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs (Reported by dtryba) |
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cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured (Reported by Tzafrir Cohen) |
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app_queue: Music On Hold for real-time queues is not reset to default (Reported by Nathan Bruning) |
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Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate (Reported by Allen Ford) |
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res_pjsip: Loss of SIP registrations causing unavailable endpoints (Reported by Richard Mudgett) |
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res_ari: Memory leaks in ARI when using Content-Type: application/json (Reported by David Hajek) |
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chan_sip: tcpbind uses wrong source address (Reported by Ksenia) |
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Dual-Stack server cannot be used as IPv4 client via TCP/TLS (Reported by Alexander Traud) |
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vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell) |
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res_pjsip_mwi: uninitialized value from ast_strings_match (Reported by Corey Farrell) |
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False positive busy checks when icalendar’s recurrence-id mechanism is involved (Reported by Benoît Dereck-Tricot) |
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Status of RFC 3323 and PJSIP (Reported by dtryba) |
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app_queue: does its check-makeannouncement-logic twice each head-caller-loop (Reported by Stefan Engström) |
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Contact is improperly translated after d178f497 (Reported by Sean Bright) |
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Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes) (Reported by Ross Beer) |
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A codeblock that maintains a bug,but maybe the codeblock will never run (Reported by Huangyx) |
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Realtime config fail with PostgreSQL version before 9.1 (Reported by Rodrigo Ramirez Norambuena) |
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bridge_native_rtp: half-way direct media when using early bridging (Reported by Jean Aunis – Prescom) |
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Crash in pubsub_on_rx_request NULL pointer – Possible PJSIP Vulnerability (Reported by Ross Beer) |
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tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) |
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SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 (Reported by Marcello Ceschia) |
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RTCP needs better packet validation to resist port scans. (Reported by Richard Mudgett) |
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RTP: One way audio with direct media and strictrtp=yes. (Reported by Richard Mudgett) |
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module reload res_calendar.so does not reload everything in calendar.conf (Reported by Jesper) |
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res_calendar does not process CalDAV from Owncloud [fix included] (Reported by Stefan Gofferje) |
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res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured. (Reported by Jesper) |
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RTP Multicast of L16 (type 10): Asterisk and wireshark disagree (Reported by Tzafrir Cohen) |
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external_media_address and external_signaling_address don’t always honor localnet (Reported by Walter Doekes) |
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CDR: CDR(start,u) function won’t work in cdr_custom config (Reported by Jacek Konieczny) |
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chan_sip: Asterisk crashing when subscription doesn’t get set (Reported by Bryan Walters) |
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res_smdi: convert to astobj2 (Reported by Corey Farrell) |
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SDP origin attribute modified when issuing re-INVITE because of directmedia=yes (Reported by saghul) |
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alembic: prune_on_boot fix erroneous (Reported by Florian Floimair) |
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When in queue on g722 with interruptions, music on hold can get stuck and no longer play (Reported by Jens T.) |
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nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) |
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PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) |
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Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive (Reported by Ross Beer) |
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Crash when freeing dtls_cfg->cafile (Reported by Richard Kenner) |
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ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen) |
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libc segfault upon entry into app_directory (Reported by David Moore) |
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Sending a “tel” uri in a From or To header in an unauthenticated message causes asterisk to crash (Reported by Ross Beer) |
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core: ast_safe_system command injection possible. (Reported by Corey Farrell) |
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res_rtp_asterisk: Media can be hijacked even with strict RTP enabled (Reported by Joshua Colp) |
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Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name (Reported by James Terhune) |
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app_minivm fails to clean up mkstemp files (Reported by Walter Doekes) |
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several filename bugs in Record() application (Reported by klaus3000) |
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alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) |
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ControlPlayback fails to play files with names containing certain non-alpha characters (Reported by Jonathan White) |
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When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. (Reported by Jim Van Meggelen) |
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When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored (Reported by Eelco Brolman) |
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app_queue: Wrong queue stat calculation (Reported by sungtae kim) |
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Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used (Reported by Torrey Searle) |
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XMPP OAuth not working due to inverted logic (Reported by Michael Kuron) |
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res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical (Reported by Mark Thompson) |
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If wget is not installed and “or” is not available, external components (excluding pjsip) are not installed (Reported by Seán C. McCord) |
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Either asterisk or pjproject isn’t re-using tcp connections (again) (Reported by George Joseph) |
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IPv6 receive address in message doesn’t include brackets (Reported by Scott Griepentrog) |
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res_rtp_asterisk: RTCP statistics are not available when native bridge is used (Reported by Torrey Searle) |
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RTP session is not fully destroyed on channel hangup (Reported by Matt Jordan) |
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Asymmetric codecs when asymmetric_rtp_codec=no (Reported by Jesse Ross) |
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Asterisk 15.0.0-Beta1 does not compile (Reported by Ira Emus) |
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res_pjsip: PJSIP presence – missing braces around the status element in XML (Reported by Abraham Liebsch) |
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Asterisk won’t compile on Fedora 26 with devmode enabled. (Reported by Corey Farrell) |
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res_pjsip: TLS connection not stable (Reported by Ian Gilmour) |
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Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly (Reported by Sergej Kasumovic) |
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say.c calls for sounds in the subdir “digits” that don’t exist (in Core). SayUnixTime or other Say… apps will fail out when they call these sounds. (Reported by Nicolas Riendeau) |
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sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 (Reported by Corey Farrell) |
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app_playback.c:say_date_generic use timezonename parameter (Reported by Holger Hans Peter Freyther) |
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res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use (Reported by Joshua Colp) |
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res_stasis_snoop: When recording a snoop channel (using ARI) where no media is being received, no recording happens when theres no media (Reported by Dan Jenkins) |
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confbridge: Name recordings are left on filesystem (Reported by Sergej Kasumovic) |
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chan_iax2: On reload MWI taskprocessors keep adding up (Reported by Sergej Kasumovic) |
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sounds: New 3-D Binaural audio features require new sound prompts (Reported by Rusty Newton) |
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French conf-adminmenu, conf-usermenu prompts differ in content from the English files (Reported by Benoit Duverger) |
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Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) (Reported by Rusty Newton) |
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configs: Erroneous load directive in sample configuration results in “Error loading module ‘res_pjsip_multihomed.so'” (Reported by HZMI8gkCvPpom0tM) |
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core: when setting ‘maxfiles’ in asterisk.conf, a message is printed, even in rasterisk -x (Reported by Tzafrir Cohen) |
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res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing ‘@’ (Reported by Maxim Vasilev) |
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res_rtp_asterisk: Deadlock when TURN session in use (Reported by Jatin Jain) |
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ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting (Reported by James Terhune) |
New Features made in this release:
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AMI : Add CancelAtxfer Action (Reported by Thomas Sevestre) |
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core: Add support for timelen parsing to ast_parse_arg and ACO. (Reported by Corey Farrell) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0-rc1
Thank you for your continued support of Asterisk!